The Third Generation Partnership Project (3GPP) Release 99 introduced carrying circuit-switched (CS) voice over a dedicated transport channel (DCH) on both the uplink (UL) and downlink (DL) in order to control the adaptive multi-rate (AMR) data rate. AMR is an audio data compression scheme optimized for voice coding.
AMR coding is used to select the optimum channel (half or full rate) and codec mode (voice and channel bit rates) to deliver the best combination of voice quality and system capacity. AMR coding improves the quality and robustness of the network connection while sacrificing some voice clarity. The AMR codec has the capability of generating voice frames containing a variable number of bits according to a set of possible data rates. The selection of a higher data rate results in a higher voice quality at the expense of requiring more resources to transmit the data.
FIG. 1 is a block diagram of an AMR voice system 100. The AMR system may include a transmit side 110 and a receive side 120. The transmit side 110 may comprise a 8-bit A law or μ-law pulse code modulator, a low pass filter, an analog-to-digital converter, a voice activity detector, a voice encoder, a comfort noise generation system, and an error concealment mechanism to combat the effects of transmission errors and lost packets. The receive side 120 may comprise components for the inverse functions.
As shown in FIG. 1, the voice encoder takes its input as a 13 bit uniform Pulse Code Modulated (PCM) signal either from the audio part of the WTRU or on the network side, from the Public Switched Telephone Network (PSTN) via an 8-bit A-law or μ-law to 13-bit uniform PCM conversion. The encoded voice at the output of the voice encoder is packetized and delivered to a discontinuous transmission control and operation block (i.e. network interface). In the receive side 120, the inverse operations take place.
The detailed mapping between input blocks of 160 voice samples in 13 bit uniform PCM format to encoded blocks (in which the number of bits depends on the presently used codec mode) and from these to output blocks of 160 reconstructed voice samples is described in 3GPP TS 26.090. The encoding scheme is Multi-Rate Algebraic Code Excited Linear Prediction. The bit-rates of the source codec are listed in Table 1.
The multi-rate voice encoder is a single integrated voice codec with eight source rates from 4.75 Kbit/s to 12.2 Kbit/s, and a low rate background noise encoding mode. The voice coder is capable of switching its bit-rate every 20 ms voice frame upon command. An AMR voice codec capable WTRU supports the following source codec bit-rates listed in Table 1.
TABLE 1Source codec bit-rates for the AMR codec.Codec modeSource codec bit-rateAMR_12.2012.20Kbit/s (GSM EFR)AMR_10.2010.20Kbit/sAMR_7.957.95Kbit/sAMR_7.407.40Kbit/s (IS-641)AMR_6.706.70Kbit/s (PDC-EFR)AMR_5.905.90Kbit/sAMR_5.155.15Kbit/sAMR_4.754.75Kbit/sAMR_SID1.80Kbit/s (see note 1)
In a 3GPP Release 99 system, when CS voice is carried over the DCH, the AMR data rate on the UL may be controlled using transport format combination (TFC) control messages transmitted by a radio network controller (RNC). The network may alleviate UL congestion by reducing the data rate of a WTRU utilizing CS voice transmission.
3GPP Release 6 introduced high-speed uplink packet access (HSUPA) to provide higher data rates for uplink transmissions. As part of HSUPA, a new transport channel, the enhanced dedicated channel (E-DCH) was introduced. The E-DCH is a transport uplink channel that is used to improve capacity, data throughput, and reduce the delays for the dedicated channels in the UL. Typically in each transmission time interval (TTI), one transport block of data may be transmitted. The size of the transport block may very for each TTI.
In HSUPA, the MAC layer may multiplex data from multiple logical channels or MAC-d flows to a single E-DCH. The network may configure which MAC-d flows may be multiplexed together and the highest priority MAC-d flow being transmitted dictates the quality of service (QoS) parameterization of a transmission. A MAC-d flow may be defined as a flow of MAC-d PDUs which belong to logical channels that share some QoS characteristics.
Support for the transport of CS voice over the High-Speed Downlink Shared Channel (HS-DSCH) and the E-DCH has been introduced in Releases 7 and 8. This feature has several benefits, such as minimizing the use of DCH transport channel in a cell and faster call setup.
Currently, there is no method describing how to control the UL data rate of a CS voice service when it is carried over the E-DCH. There exists a need to implement the rate control of a CS voice carried over the E-DCH.
In a 3GPP Release 6 system, the E-DCH uses different scheduling mechanisms and a hybrid automatic repeat request (HARQ). The scheduling is typically based on scheduling grants sent by a Node-B scheduler to control the WTRU's uplink transmissions. The WTRU may transmit scheduling information to request additional resources. Scheduling grants may include absolute grants and relative grants. Absolute grants set an absolute value of an upper limit of the power a terminal may use for a transmission. Relative grants meanwhile update the resource allocation by indicating a value such as “up”, “down”, or “hold”. However, a clear mapping between transport formats and supportable data rates does not exist. Further, the resource allocation tasks of the E-DCH are shared between the RNC and the Node-B. Additionally, a WTRU may need to reduce its UL data rate due to transmission power limitations at a cell edge.
Accordingly, there exists a need to control the AMR data rate when CS voice service is transmitted over the E-DCH. By controlling the AMR data rate when CS voice is transmitted over the E-DCH, the UL voice coverage may be extended.